[Asterisk-Users] Need help with two-stage ringing macro
I've been using the following macro to ring SIP and IAX devices for a
few seconds, and then add on a cell phone if there is no answer on the
SIP or IAX device. Periodic problems began a few versions ago and now
the problem happens every time with 1.2.9 and 1.2.9.1.
The problem is that when a call from the PRI falls through to voicemail,
the call is dropped before the voicemail greeting is heard. Debug shows
that voicemail is starting and that Asterisk is dropping the call on the
PRI. Calls made from SIP or IAX devices work fine.
[macro-followme]
;
; modified standard extension macro for two-stage ringing.
;
; It will call the destinations in ${ARG4} for ${ARG2} seconds, and
; if that fails, the destinations in ${ARG5} for ${ARG3} seconds. If
; that also fails, it will send the call to voice mail for extension
; ${ARG1}.
;
; Note: if you want it to ring phone1 first, then phone1 AND phone2
; next, you have to list phone1 in both lists. Otherwise it will
; stop ringing on phone1.
;
; ${ARG1} - voice mail context
; ${ARG2} - Extension
; ${ARG3} - Time to ring stage 1
; ${ARG4} - Time to ring state 1 + 2
; ${ARG5} - Device(s) to ring stage 1
; ${ARG6} - Device(s) to ring stage 2
;
exten = s,1,SetCallerID(${CALLERIDNUM:-10:10}) ; Send only the last 10
digits
exten = s,2,NoOp(CallerID After:${CALLERIDNUM})
exten = s,3,SetAccount(${ARG2})
exten = s,4,Dial(${ARG5},${ARG3},rt) ; Ring the primary group
exten = s,5,Dial(${ARG5}${ARG6},${ARG4},rt) ; Add in the secondary group
exten = s,6,Voicemail([EMAIL PROTECTED]) ; send to vm as unavail
exten = s,7,Hangup
exten = s,106,Voicemail([EMAIL PROTECTED]) ; send to vm w/ busy
announce
exten = s,107,Hangup
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[Asterisk-Users] pickup problem
Hi! Could somebody help me with pickup feature? I've set callgroup = 1 pickupgroup = 1 for my phones in sip.conf, but if I try to pickup call with *8 asterisk output to console Jun 6 15:04:44 WARNING[11857]: pbx.c:2401 __ast_pbx_run: Invalid extension '*', but no rule 'i' in context 'office' Thanks! -- DSS5-RIPE DSS-RIPN 2:550/[EMAIL PROTECTED] 2:550/[EMAIL PROTECTED] xmpp:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://neva.vlink.ru/~dsh/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] skype out
On Tuesday, June 06, 2006 12:10 AM Andrei (MPI) wrote: I'm using SIP-to-Skype/Skype-To-SIP software gateway called Uplink (found in Wiki): http://nch.com.au/skypetosip/ - which is free and working great so far. Downsides are: Only that it produced not RFC conform SIP headers which are blocked by some firewalls (e.g. Juniper). I tried to contact their technical support several times but have not received any answer whatsoever. Very poor support Kind regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP!!!! Weird TDM2406E unable to bridge all outgoing calls.
Hi all, I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in my office. the out going calls symptom like when called party pickup the phone but the calling party still hearing the ring tone from the IP phone. Please light me up. it been many sleepless night by googling around trying to get the right answers. The digium card running on Intel 915G chipset. Below are my zaptel configurations. asterisk version: 1.2.9.1 Zaptel Version: 1.2.6 Echo Canceller: KB1 # Span 1: WCTDM/0 Wildcard TDM2400P Prototype Board 1 fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 | | fxsks=21 fxsks=22 fxsks=23 fxsks=24 # Global data loadzone= us defaultzone = us [root#] lspci -vb 06:00.0 Ethernet controller: Digium, Inc. Wildcard TDM2400P (rev 11) Subsystem: Digium, Inc. Wildcard TDM2400P Flags: bus master, medium devsel, latency 32, IRQ 3 I/O ports at b800 Memory at ff52 (32-bit, non-prefetchable) Expansion ROM at ff50 [disabled] Capabilities: [c0] Power Management version 2 [root#] dmesg snip --- Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.6 Echo Canceller: KB1 PCI: Found IRQ 3 for device :06:00.0 PCI Config reg is 02900117 WCTDM2400P: New Reg: fe59! Detected REG0: 0100 Detected REG1: 7849 Detected REG2: 001d (pre) Reg fc is 5027 (post) Reg fc is 5024 Detected REG2: wctdm2400p: reg is a04c0004 Resetting the modules... During Resetting the modules... After resetting the modules... Port 1: Installed -- AUTO FXO (MALAYSIA mode) Port 2: Installed -- AUTO FXO (MALAYSIA mode) Port 3: Installed -- AUTO FXO (MALAYSIA mode) Port 4: Installed -- AUTO FXO (MALAYSIA mode) Port 5: Installed -- AUTO FXO (MALAYSIA mode) Port 6: Installed -- AUTO FXO (MALAYSIA mode) Port 7: Installed -- AUTO FXO (MALAYSIA mode) Port 8: Installed -- AUTO FXO (MALAYSIA mode) Port 9: Installed -- AUTO FXO (MALAYSIA mode) Port 10: Installed -- AUTO FXO (MALAYSIA mode) Port 11: Installed -- AUTO FXO (MALAYSIA mode) Port 12: Installed -- AUTO FXO (MALAYSIA mode) Port 13: Installed -- AUTO FXO (MALAYSIA mode) Port 14: Installed -- AUTO FXO (MALAYSIA mode) Port 15: Installed -- AUTO FXO (MALAYSIA mode) Port 16: Installed -- AUTO FXO (MALAYSIA mode) Port 17: Installed -- AUTO FXO (MALAYSIA mode) Port 18: Installed -- AUTO FXO (MALAYSIA mode) Port 19: Installed -- AUTO FXO (MALAYSIA mode) Port 20: Installed -- AUTO FXO (MALAYSIA mode) Port 21: Installed -- AUTO FXO (MALAYSIA mode) Port 22: Installed -- AUTO FXO (MALAYSIA mode) Port 23: Installed -- AUTO FXO (MALAYSIA mode) Port 24: Installed -- AUTO FXO (MALAYSIA mode) VPM Revision: 01 VPM: U-law mode VPM: DTMF threshold set to 1250 VPM: Present and operational (Rev B) Found a Wildcard TDM: Wildcard TDM2400P Prototype (24 modules) Registered tone zone 0 (United States / North America) --- snip --- [root]# cat zapata.conf [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no useincomingcalleridonzaptransfer=yes busycount=4 callprogress = yes ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 ;Include AMP configs #include zapata_additional.conf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] syslog server
I am using syslog-ng, with mysql, and php-syslog-ng, so you get a web interface to search for logs, and a huge capacity on the mysql databse, I have a syslog-ng with the above configuration, and is handleing 5 million syslog message per day. On 6/6/06, Matthew Warren [EMAIL PROTECTED] wrote: Does anyone know a good syslog server to use for grandstream phones?I wantto set this up to see what is happening with the grandstreams.Easy andFree preferably.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI (I.430) over ethernet
2006/6/7, James Harper [EMAIL PROTECTED]: I've thought of this before, but my idea was to have a small box (aboutthe size of an nt1) with an S/T interface on one side and ethernet onthe other. A SBC with built in Ethernet and a minipci slot might do, but a dedicated device should be able to be mass produced pretty cheaplythough (~$100 range).The other problem I have is the lack of cheap ISDN adapters inAustralia. I have only found 1 4HFC adapter here and couldn't make it work properly.Do you mean using a minipci ISDN adapter like http://www.junghanns.net/en/quadBRImini_produkt.html ? I think a $50 target for such a module (or a 1 ISDN port one) would be difficult to achieve.Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STNU spport
hi, actully, i need asterisk as a stun client... have any idea On 6/6/06, unplug [EMAIL PROTECTED] wrote: HI, There is a parameter NAT can be set in the configuration file. Is it the way that we can use to support NAT by setting nat=yes in the file instead using other NAT resolving tools like stun? On 6/6/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Tue, 2006-06-06 at 16:12 +0800, Chen Fan wrote: hi, We need STUN client support for asterisk... becasue the service provider only offer STUN interface,, so i can not connect asterisk to their server all stun does is resolve your external IP by sending data to a foreign server which looks at the IP and returns it back to you. It has nothing to do with the channel used other than SIP will then use that IP (which can be defined by either externhost or externip - dont forget localnet too in sip.conf). i have found that there someone is develop res_stun.c ..but still not release... likely that is just going to replace the externip value in the chan_sip driver. I cant imagine that it would do much more than that. Have you set both externip and localnet in sip.conf and checked to see if that works? If you dont do NAT on your end it wont even be required. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQBEhUFA+1olxlzQw5cRAoDtAKCK5ufDIpmsXG/p2ydcj3VDqxA7jgCcCAHi bpFsVQ8FJuxF+crAEm2hwZE= =VQtX -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeffery `∧ ∧︵ ミ^r^ミ灬)~ iaxtel Num: 1-700-576-1311 fwdnet Num: 728150 http://www.diaip.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : Re: [Asterisk-Users] asterisk-1.2.9 / res_snmp.so
--- [EMAIL PROTECTED] a écrit : hello, How asterisk could support res_snmp even this module don't help to monitor all asterisk features? monitoring asterisk with snmp would be a good thing. Which solution ? Harry --- Kristian Kielhofner [EMAIL PROTECTED] a écrit : [EMAIL PROTECTED] wrote: I upgrade 1.2.7-1 to 1.2.9 but asterisk is not stable I 've lost call SIP-ZAP. channels. i can't hear sound because of res_snmp.so . Is it a bêta release ?? I downgrade to 1.2.8 or 1.2.7 I do hope 1.4 will be a real stable realease Harry Harry, res_snmp.so is not in 1.2.9. I don't know what version you are running, but either it isn't 1.2.9 or you added res_snmp on your own. That could explain some of your other issues. -- Kristian Kielhofner __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing call bridging
On 6 Jun 2006, at 05:05, [EMAIL PROTECTED] wrote:
Thanks for the info. It would be an external program. I have been
looking at the originate manager command, but it looks like it
would not bridge 2 external numbers. One of the number has to be a
local extension. Suppose I want it to dial 2 number on zap and then
bridge them, I probably would still have to do a dialplan gimmick,
is that right?
Well not really gimmick, originate has done the hard part.
So, lets say you want to call
555-666-777 and bridge it with a call to 111-222-333
you'd put something like this in extensions.conf:
[globals]
TRUNK=iax/myprovider
[out-bridge]
exten = _[0-9].,1,Dial(${TRUNK}/${EXTEN})
Then you have originate make a call to iax/myprovider/555666777
and pass it
context = 'out-bridge' exten = '111222333' priority = 1
If you use the same provider and technology for both legs
of the call, you may get lucky and your asterisk can
re-invite the call, so that it is no-longer in the
media path.
Then again, if you get unlucky the provider's billing software
may get confused by this.
Tim.
Tim Panton
[EMAIL PROTECTED]
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[Asterisk-Users] CISCO POE
Hi do you know how to make a cable for powering a POE Cisco Phone from an not cisco POE Switch ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
Erick Baum wrote: The worst ongoing issue has been the echo and the really crappy speakerphone. The customer is pretty much used to it now. But we're slowly replacing them with Polycom's as new people come on and as others just get fed up. Unfortunately one of the phones met it's doom by way of a hammer. But I guess, what do you expect for under a hundred bucks. Wow, I nearly bought some of these, but since the customer wouldn't pay that much ended up getting some £30 chinese phones instead (not quite as good spec. but sounds like they work at least as well). Had no problems with the 2 at home, and so far (touch wood) the other 18 haven't had any major problems. Mind you no-one uses the speaker phone, now if only I could get a headset for them. Erick On 6/6/06, *Daniel Salama* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I enabled call-waiting from the tftp configuration and it now works. What firmware are you using and where can I get it? My client complaints that the phone stops working every once in a while with no explanation. My client says that he could be using the phone with no problem and a few minutes later, when he wants to make a call, the phone will always give a fast busy after pressing the fourth digit. My workaround to him was to reboot the phone. That seems to solve the problem, however, it's not practical to have that problem in an office environment with 18 GXP-2000. Any ideas what the problem could be? Thanks, Daniel On Jun 6, 2006, at 6:26 PM, Mike wrote: I can't say why you're having this problem, but I can tell you that my phone can receive (and make) multiple calls easily. It might have more to do with Asterisk than the GXP2000. I am using the latest release firmware, not a beta. Mike -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Daniel Salama Sent: June 6, 2006 4:12 PM To: Non-Commercial Discussion Asterisk Subject: [Asterisk-Users] GXP-2000 I'm using a few GXP-2000 with firmware *MailScanner warning: numerical links are often malicious:* 1.0.2.13 http://1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I see Got SIP response 486 Busy back on the console. Is there a way to make the phone receive calls on all 4 lines? 2) Is there any more documentation as to the tftp configuration file? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk load balancing setup
Hi all, I would like to setup a redundancy/load balancing of an asterisk system as follow. Internet DNS asterisk1 -- mysql DB +---asterisk2 --+ In such case, all user account, dial plan and other necessary information is stored in a single DB or a cluster of DB. DNS with round robin enabled to distribute the traffic coming from outside. I am not sure it is the right way to provide load balance of asterisk. Anyone can give me some advices? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
I am running 1.1.0.13 and there are no issues which are causing a problem for us. The speakerphone is not much use but we can live with that. 1.0.1.9 would stop registering after a while causing incoming calls to go straight to voicemail. 1.0.2.13 fixed this but had a bug where sometimes reviewing the missed call list caused the phone to crash. We have 35 handsets in use. On Tue, 2006-06-06 at 21:11, Daniel Salama wrote: I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I see Got SIP response 486 Busy back on the console. Is there a way to make the phone receive calls on all 4 lines? 2) Is there any more documentation as to the tftp configuration file? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : Re: [Asterisk-Users] asterisk-1.2.9 / res_snmp.so
On Wed, Jun 07, 2006 at 10:10:16AM +0200, [EMAIL PROTECTED] wrote: hello, How asterisk could support res_snmp even this module don't help to monitor all asterisk features? monitoring asterisk with snmp would be a good thing. Which solution ? Harry --- Kristian Kielhofner [EMAIL PROTECTED] a écrit : [EMAIL PROTECTED] wrote: I upgrade 1.2.7-1 to 1.2.9 but asterisk is not stable I 've lost call SIP-ZAP. channels. i can't hear sound because of res_snmp.so . Is it a bêta release ?? I downgrade to 1.2.8 or 1.2.7 I do hope 1.4 will be a real stable realease Harry Harry, res_snmp.so is not in 1.2.9. I don't know what version you are running, but either it isn't 1.2.9 or you added res_snmp on your own. That could explain some of your other issues. How asterisk could support res_snmp even this module don't help to monitor all asterisk features? monitoring asterisk with snmp would be a good thing. Which solution ? Solution to what? Could you please describe the symptoms of your problems? What makes you think that res_snmp.so is related to those symptoms? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 channel problems
Hello List We are a VoIP telco, running Asterisk. We have been having problems with our IAX2 channels for some time now. Our problems are jitter, and lost packets, resulting in bad audio quality. The weird thing is, that this mostly occurs on our local network. We have tested the network with pinging an hour, without any lost packets. One of our customers also has problems using IAX2, and he is only two networks away, according to traceroute. He is on a 100mbit dedicated connection. Is there a general problem in the IAX2 channel, which causes jitter? We are running 1.2.9.1, and have tried 1.2.7.1, 1.2.5 and 1.2.0, and we have the same problems with all of them. Our average system load is around 2-3, and we have 905 registered sip users, and around 60 calls running at all time, to queues, SIP and Zap channels. Regards Jon Schoepzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [asterisk-biz] UK Male English Voices
Yet another set? I get about 50 downloads a week for mine. Mark On Tue, 2006-06-06 at 22:27 +0100, Steve Kennedy wrote: I'd like to announce that the UK Male English Voices are now up on http://www.tel.net/ There's a complete set of base sounds and additional sounds (it should be complete compared to current Asterisk and Asterisk-Sounds-1.2.1). There's also a set with the word 'pound' replaced by 'hash' for both the base and additional sounds (only the actual replacements not a complete set). There's sets of gsm and pcm files. I'd like to thanks Jay Benham [EMAIL PROTECTED] who did all the hard work of recording them, and Jim Credland [EMAIL PROTECTED] for doing all the converting/sound work. Regards Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 channel problems
Jon Schøpzinsky ha scritto: We have been having problems with our IAX2 channels for some time now. Our problems are jitter, and lost packets, resulting in bad audio quality. The weird thing is, that this mostly occurs on our local network. We have tested the network with pinging an hour, without any lost packets. One of our customers also has problems using IAX2, and he is only two networks away, according to traceroute. He is on a 100mbit dedicated connection. Is there a general problem in the IAX2 channel, which causes jitter? We are running 1.2.9.1, and have tried 1.2.7.1, 1.2.5 and 1.2.0, and we have the same problems with all of them. Our average system load is around 2-3, and we have 905 registered sip users, and around 60 calls running at all time, to queues, SIP and Zap channels. On our system when the load is around 3 we lose packet in iax, but due to excessive load, simply decreasing the number of calls solves the problem. fastagi and load balancing greatly mitigated the problem. Also forcing jitterbuffer on machines not at the edges was a problem. I leave it enabled only on the machines with digium cards and ata connected. I also use sip when possbile, a lot of people from previous posts reports sip being better than iax when jitter comes to play, but in my experience 1.2.7 works quite well ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delay on calls
Hi, I have a 1.2.4 * box with two HFC modems using chan_modem_i4l and several SIP phones and ATA's. We have a terrible delay on calls between the PSTN (isdn BRI) and the SIP phones. All internal calls are fine. My first thought was that the transcoding could cause the delay but all of the SIP phones default to ulaw so there should not be any transcoding needed. I also checked the load on the server and it is well below 10% cpu utilisation and load average of below 1. The same setup on two other servers works fine. I do not even know where to start looking - any suggestions will be appreciated. Thank you Marnus van Niekerk -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] a new asterisk version
Hi All, I need a suggestion. I want to run only IAX on two windows based PCs and asterisk Can you suggest which asterisk, libpri and zaptel versions should i use? do i need some othermodules also? Also where will i find the guide to compile astreisk Actually i have installed,comnpiled and used astreisk-1.0.3 on Red hat 9 which was not that stable. Now i have Red hat Enterprise on my PC. ithink there are newer stable versions which can run on Redhat Enterprise Linux. Kindly help, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a new asterisk version
http://www.asterisk.org/download http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+CentOS amna saleem wrote: Hi All, I need a suggestion. I want to run only IAX on two windows based PCs and asterisk Can you suggest which asterisk , libpri and zaptel versions should i use? do i need some other modules also? Also where will i find the guide to compile astreisk Actually i have installed,comnpiled and used astreisk-1.0.3 on Red hat 9 which was not that stable. Now i have Red hat Enterprise on my PC. i think there are newer stable versions which can run on Redhat Enterprise Linux. Kindly help, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I can hear only one way when I use nokia e-60 with X-lite
Hi I am facing some problems in making calls to Nokia E60 ,from other sip extensions, I am able to hear clearly when I use the X-lite clients , but on Nokia E60 , I cannot hear anything ,ie whenever a call is made , the user who uses X-lite hears everything what the Nokia user says , but Nokai user cannot hear anything at all Please advice me , where I should check , the problem , is it because of codec selection , I did try with other codecs like ulaw , the experience was same I am using asterisk 1.2.8 on RHEL4 Thanks Joseph John my sip.conf contains [666] ; Xlite Phone username=666 type=friend secret=666 ;qualify=no ;port=5060 ;notransfer=yes host=dynamic context=from-internal disallow=all allow=alaw [221] ;; Nokia E-60 username=221 type=friend secret=221 ;qualify=no ;port=5060 ;notransfer=yes host=dynamic context=from-internal disallow=all allow=alaw Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
I have a client who has about six of these phones. Luckily (for me, not for them) they were purchased before I came into the picture. Daniel Salama wrote: I have heard complaints from my client about the speakerphone and they are now You don't notice any problems when using the speaker-phone, but the person on the other end hears echo, and quite a lot of it. , I guess, getting used to picking up the handset :). My client uses them exclusively with headsets (in a call center) so the quality of the speaker-phone isn't an issue for them. I have heard any echo problems so far. What bothers me the most is that the phone stops working often (multiple times per day). By this I mean that my client won't be able to dial anything successfully. As soon as 3 or 4 digits are entered, they get a fast busy. To solve it, they need to reboot it. It sounds as if these phones were running Windows instead of Linux :) Do you have multiple phones going down at the same time? If so, monitor them with qualify=500 in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. Also I have heard some phones have trouble with broadcast packets (at least this has been said about the spa-841 on the wiki). You should strongly consider putting them on a separate vlan to avoid any issues like that. In the future, for phones under $100 then look at the spa-841 phones. Anyway, what firmware did you use that solved so many of your problems? http://www.voip-info.org/wiki/view/GXP-2000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] I can hear only one way when I use nokia e-60 withX-lite
Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af John Joseph Sendt: 7. juni 2006 13:59 Til: Asterisk Users Emne: [Asterisk-Users] I can hear only one way when I use nokia e-60 withX-lite Hi I am facing some problems in making calls to Nokia E60 ,from other sip extensions, I am able to hear clearly when I use the X-lite clients , but on Nokia E60 , I cannot hear anything ,ie whenever a call is made , the user who uses X-lite hears everything what the Nokia user says , but Nokai user cannot hear anything at all Please advice me , where I should check , the problem , is it because of codec selection , I did try with other codecs like ulaw , the experience was same I am using asterisk 1.2.8 on RHEL4 Thanks Joseph John my sip.conf contains [666] ; Xlite Phone username=666 type=friend secret=666 ;qualify=no ;port=5060 ;notransfer=yes host=dynamic context=from-internal disallow=all allow=alaw [221] ;; Nokia E-60 username=221 type=friend secret=221 ;qualify=no ;port=5060 ;notransfer=yes host=dynamic context=from-internal disallow=all allow=alaw Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: GXP-2000 (steer clear)
On Tue, Jun 06, 2006 at 11:26:20PM -0400, Daniel Salama wrote: Well, these are encouraging words :) You're basically telling me that I should tell my client to buy other phones. I agree that you cannot compare these phones with Cisco or Polycom. After all, like you said, what do you expect for under $90. However, the fact is that my client just recently invested in these and it will be hard, if not impossible, for me to tell my client to swap them for Polycoms or something else at a much higher cost. I have heard complaints from my client about the speakerphone and they are now, I guess, getting used to picking up the handset :). I have heard any echo problems so far. What bothers me the most is that the phone stops working often (multiple times per day). By this I mean that my client won't be able to dial anything successfully. As soon as 3 or 4 digits are entered, they get a fast busy. To solve it, they need to reboot it. It sounds as if these phones were running Windows instead of Linux :) Anyway, what firmware did you use that solved so many of your problems? I've only had bad experiences with these phones and steer clear of them. In the same price range you can now get the Thomson ST-2030 or Polycom 430 for a much, much better user experience. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CLI comand to register softphones without close them:
Hi;I've a question: I use asterisk -R so I can see what's appening in my asterisk and the session of the calls: I use the vrrp protocol, I use 2 asterisk box;when the master falls down, the slave goes up, and I use X-lite,Phoner,3CXphone;some of this softphones are immediately registered to the slave, but sometimes this don't happen;Imust close the softphone from my xp and restart them,and then in the CLI interface I can see that the softphone are restistered in this way: Verbosity is at least 3 -- Registered SIP '651' at 192.168.251.10 port 5060 expires 900 -- Registered SIP '650' at 192.168.251.10 port 5061 expires 1800 -- Registered SIP '655' at 192.168.251.10 port 3571 expires 900 and then they are ok and I can call. IfI use the restart now command or reload in the CLI of the slave, asterisk don't see that the softphones are up, I must close the softphones and restart them. How can I reload the softphones, without restart them? THANKS, Emanuele ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Cellular boosters
This isn't really a fix for the people missing calls, but one solution I found was to limit the amount of time a call rings for to a cell phone. If it doesn't answer in X seconds, then dial again. This isn't a perfect solution, but helps some. On 6/6/06, Colin Anderson [EMAIL PROTECTED] wrote: We use Motorola v551's as extensions on our Asterisk system with ahomebrew find me/follow me dialplan. It works great except where coverage ispoor then of course the inbound call hits voicemail. This has nothing to do with Asterisk and everything to do with our cellular provider, but since youguys are telephony pros I'd like to ask if anyone has had any good or badexperience with gain boosters for cells from those snake oil stick on things all the way up to powered one-watt boosters. Ideally, I'd like a situationwhere I replace the stock OEM antenna with something else for $10 and away Igo. I have a hundred guys with v551s that are pissed about missed calls, so any and all suggestions are welcome. tia___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GXP-2000 (steer clear)
On Wed, June 7, 2006 14:09, Louis-David Mitterrand said: On Tue, Jun 06, 2006 at 11:26:20PM -0400, Daniel Salama wrote: Well, these are encouraging words :) You're basically telling me that I should tell my client to buy other phones. I agree that you cannot compare these phones with Cisco or Polycom. After all, like you said, what do you expect for under $90. However, the fact is that my client just recently invested in these and it will be hard, if not impossible, for me to tell my client to swap them for Polycoms or something else at a much higher cost. I have heard complaints from my client about the speakerphone and they are now, I guess, getting used to picking up the handset :). I have heard any echo problems so far. What bothers me the most is that the phone stops working often (multiple times per day). By this I mean that my client won't be able to dial anything successfully. As soon as 3 or 4 digits are entered, they get a fast busy. To solve it, they need to reboot it. It sounds as if these phones were running Windows instead of Linux :) Anyway, what firmware did you use that solved so many of your problems? I've only had bad experiences with these phones and steer clear of them. In the same price range you can now get the Thomson ST-2030 or Polycom 430 for a much, much better user experience. Where do you purchase the Thomson or Polycoms for a comparable price as the GXP2000? I'd like to buy an ST2030 or 430 for under EUR 90 myself too! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GXP-2000 (steer clear)
While I would agree with you, the price difference between a GXP-2000 and a Polycom 430 or a Thomson ST-2030. These latter units are, at least, twice as expensive as the GXP-2000.BTW, I never heard of the Thomson ST-2030, but it looks _really_ nice.Thanks,DanielOn Jun 7, 2006, at 8:09 AM, Louis-David Mitterrand wrote:I've only had bad experiences with these phones and steer clear of them. In the same price range you can now get the Thomson ST-2030 or Polycom 430 for a much, much better user experience. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CISCO POE
A schema for the RJ45 cable pinouts to power a Cisco phones from a non-Cisco switch can be found on the WIKI here http://www.voip-info.org/wiki/index.php?page=Cisco+POE Another option is to use the PowerSense BL-8858-01 PoE Converter, which converts IEEE 802.3AF to Cisco CDP, and will run you about $20 per module. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: nik600 [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 4:37 AM Subject: [Asterisk-Users] CISCO POE Hi do you know how to make a cable for powering a POE Cisco Phone from an not cisco POE Switch ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
They don't all go down at the same time, or at least, my client hasn't noticed. I just added the qualify option. Let's see how that goes.As for the SPA-841, I have a client with a few of them and he cannot stop complaining about the bad audio quality. I replace a couple with a PAP-2 and another one with the GXP-2000 and he claims the quality to be incredibly better for both the PAP2 and the GXP-2000. He hasn't complained about the problems I mentioned on the GXP-2000 - yet :)Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time? If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. Also I have heard some phones have trouble with broadcast packets (at least this has been said about the spa-841 on the wiki). You should strongly consider putting them on a separate vlan to avoid any issues like that. In the future, for phones under $100 then look at the spa-841 phones. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SpeedTouch 780WL
Has anyone had any experience with this router?? I am looking to use it because I want to use a DECT phone in conjunction with VoIP and this seems to check all the boxes for Wi-Fi, ADSL and VoIP all at a good price.. I have never used Speedtouch hardware before so any feedback would be great.. TIA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: GXP-2000 (steer clear)
On Wed, Jun 07, 2006 at 08:27:28AM -0400, Daniel Salama wrote: While I would agree with you, the price difference between a GXP-2000 and a Polycom 430 or a Thomson ST-2030. These latter units are, at least, twice as expensive as the GXP-2000. BTW, I never heard of the Thomson ST-2030, but it looks _really_ nice. I get the ST-2030 from a french reseller for ~ 95 EUR/unit. The Polycom IP430 is more in the 140 EUR range however, but it has a real speakerphone and integrated POE (unlike the IP300). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] regexp issue
Hello to all
I had Asterisk dialing the PSTN through a defined trunk.
But when I enabled the SIP URI calls Asterisk stopped contacting
the PSTN trunk
The SIP URI dial code (who created the problem) is this:
exten = _.,1,NoOp(Incoming Call from from-internal-custom extension
${CALLERID} for [EMAIL PROTECTED])
exten = _.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _.,5,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _.,6,NoOp(@${SIPDOMAIN} is remote, forwarding...)
exten = _.,7,Macro(uridial,[EMAIL PROTECTED])
exten = _.,8,HangUp()
exten = _.,10,Goto(custom-noturi,${EXTEN},1)
exten = h,1,HangUp()
How can I say that this code is just for calls to foreign domains?
Something like:if (SIPDOMAIN != fccn.pt)
Regards
Joao Pereira
___
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Re: SV: [Asterisk-Users] I can hear only one way when I use nokia e-60 withX-lite
Hi Jon Thanks for the mail I am just checking NokaiE60 and E61 as PBX client only , right now , the NAT issue does not arise for my problem --- Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af John Joseph Sendt: 7. juni 2006 13:59 Til: Asterisk Users Emne: [Asterisk-Users] I can hear only one way when I use nokia e-60 withX-lite Hi I am facing some problems in making calls to Nokia E60 ,from other sip extensions, I am able to hear clearly when I use the X-lite clients , but on Nokia E60 , I cannot hear anything ,ie whenever a call is made , the user who uses X-lite hears everything what the Nokia user says , but Nokai user cannot hear anything at all Please advice me , where I should check , the problem , is it because of codec selection , I did try with other codecs like ulaw , the experience was same I am using asterisk 1.2.8 on RHEL4 Thanks Joseph John my sip.conf contains [666] ; Xlite Phone username=666 type=friend secret=666 ;qualify=no ;port=5060 ;notransfer=yes host=dynamic context=from-internal disallow=all allow=alaw [221] ;; Nokia E-60 username=221 type=friend secret=221 ;qualify=no ;port=5060 ;notransfer=yes host=dynamic context=from-internal disallow=all allow=alaw Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quad T1 Card
Ok... I am reluctant to ask this question as I believe that it may be like asking what someones favorite linux distribution is... but I need to make an informed decision. We are getting ready to upgrade from a TE210P to a quad T1 card with echo cancellation. I am trying to decide between the Sangoma card and the Digium card. I need this to have great quality and I need it to work well. I would like to hear about personal experiences and any other technical differences between the card. Again this is not intended to start a pissing contest or flame war ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: GXP-2000 (steer clear)
Any chance of the resellers details ? fadge -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Louis-David Mitterrand Sent: 07 June 2006 13:36 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: GXP-2000 (steer clear) On Wed, Jun 07, 2006 at 08:27:28AM -0400, Daniel Salama wrote: While I would agree with you, the price difference between a GXP-2000 and a Polycom 430 or a Thomson ST-2030. These latter units are, at least, twice as expensive as the GXP-2000. BTW, I never heard of the Thomson ST-2030, but it looks _really_ nice. I get the ST-2030 from a french reseller for ~ 95 EUR/unit. The Polycom IP430 is more in the 140 EUR range however, but it has a real speakerphone and integrated POE (unlike the IP300). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voipbuster dtmf tones?
I failed to transmit dtmf via voipbuster to the destination. Does anybody have success, if how to set it up? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ST-2030 reseller (was: Re: GXP-2000 (steer clear))
On Wed, Jun 07, 2006 at 01:55:04PM +0100, asterisk wrote: Any chance of the resellers details ? For the ST-2030 I use this reseller: http://www.hl2d.com Sales contact: Jehan-Philippe Le Roy Responsable des Ventes Partenaires [EMAIL PROTECTED] Tel: +33 1 39 51 60 32 Fax: +33 1 39 51 86 91 49 rue Lamartine 78000 Versailles France, fadge -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Louis-David Mitterrand Sent: 07 June 2006 13:36 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: GXP-2000 (steer clear) On Wed, Jun 07, 2006 at 08:27:28AM -0400, Daniel Salama wrote: While I would agree with you, the price difference between a GXP-2000 and a Polycom 430 or a Thomson ST-2030. These latter units are, at least, twice as expensive as the GXP-2000. BTW, I never heard of the Thomson ST-2030, but it looks _really_ nice. I get the ST-2030 from a french reseller for ~ 95 EUR/unit. The Polycom IP430 is more in the 140 EUR range however, but it has a real speakerphone and integrated POE (unlike the IP300). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
On Wed, Jun 07, 2006 at 08:53:27AM -0400, Sean Cook wrote: Ok... I am reluctant to ask this question as I believe that it may be like asking what someones favorite linux distribution is... but I need to make an informed decision. We are getting ready to upgrade from a TE210P to a quad T1 card with echo cancellation. I am trying to decide between the Sangoma card and the Digium card. I need this to have great quality and I need it to work well. I would like to hear about personal experiences and any other technical differences between the card. Again this is not intended to start a pissing contest or flame war I'll hijack your thread for a slightly related question: there used to be a Debian package (in Woody) to install Sangoma cards. That package was called wanpipe. Now I can't find any existing Sangoma drivers in the form of standard debs. As I don't have the hardware to test this myself, I don't really bother. But if anybody just needs help with packaging, I'd be glad to lend a hand. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
Sean Cook wrote: Ok... I am reluctant to ask this question as I believe that it may be like asking what someones favorite linux distribution is... but I need to make an informed decision. We are getting ready to upgrade from a TE210P to a quad T1 card with echo cancellation. I am trying to decide between the Sangoma card and the Digium card. I need this to have great quality and I need it to work well. I would like to hear about personal experiences and any other technical differences between the card. Again this is not intended to start a pissing contest or flame war One of the primary differences between the two cards is the Sangoma h/w echo canceler handles more cases of echo then do the Digium cards. Whether you need that additional coverage is 100% dependent on your specific implementation (eg, your T1/PRI provider), and not on what the list thinks about the two products. Since there are no affordable tools to truly quantify echo for each specific implementation, as a pbx engineer your toolkit should probably include both cards. Sort of like try the less expensive card and if it doesn't address your echo issues, then try the more expensive one. The downside to using Sangoma cards is that every time you upgrade zaptel you need to reapply the Sangoma patches using their less then straight forward documentation. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
One of the primary differences between the two cards is the Sangoma h/w echo canceler handles more cases of echo then do the Digium cards. Whether you need that additional coverage is 100% dependent on your specific implementation (eg, your T1/PRI provider), and not on what the list thinks about the two products. Since there are no affordable tools to truly quantify echo for each specific implementation, as a pbx engineer your toolkit should probably include both cards. Sort of like try the less expensive card and if it doesn't address your echo issues, then try the more expensive one. No offense but isn't that like saying Don't take what the list has to say about your purchase... instead you should guess and hope you get the right answer... but if you don't, gamble again and buy two cards? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CISCO POE
many thanks for your reply i've tried to make a cable with that configuration but it seems that it doesn't work... i'm using a 7905G Cisco ip phone and an ALL0484 Switch POE thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Notice Question
I get the following * Notice ocassionally and I was curious what it means and if it can safetly be ignored or corrected. Jun 7 05:40:47 NOTICE[32153]: res_musiconhold.c:511 monmp3thread: Request to schedule in the past?!?! Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] directory
Please can any one help me how to make directories at [EMAIL PROTECTED] To use it from the ivr *411 Thanks * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
Sean Cook wrote: One of the primary differences between the two cards is the Sangoma h/w echo canceler handles more cases of echo then do the Digium cards. Whether you need that additional coverage is 100% dependent on your specific implementation (eg, your T1/PRI provider), and not on what the list thinks about the two products. Since there are no affordable tools to truly quantify echo for each specific implementation, as a pbx engineer your toolkit should probably include both cards. Sort of like try the less expensive card and if it doesn't address your echo issues, then try the more expensive one. No offense but isn't that like saying Don't take what the list has to say about your purchase... instead you should guess and hope you get the right answer... but if you don't, gamble again and buy two cards? The list cannot guess at what level of echo you are going to incur, therefore there is no way for anyone to accurately tell you how to address issues. Both cards are quality products, but with slightly different operational characteristics. If you can't afford to purchase both cards, then a safe bet is to simply purchase the Sangoma card since it can address more echo issues then the Digium card. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music On Hold not working with new 1.2.7.1 install
I have followed the instructions provided at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conf including installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now. However, no sound is heard and I get this message from the CLI when accessing MOH: -- Started music on hold, class 'default', on channel 'Zap/19-1' -- Stoped music on hold on Zap/19-1 This happens whether it's a parked call or whether I access MOH directly via: exten = 800,1,Answer exten = 800,2,MusicOnHold() Any help would be greatly appreciated. Thank you very much. Richard __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] directory
>From the IVR someone will usually be dialing #, not *411 (although I suppose both work). In AAH, the directories are setup automatically when you setup your extensions. Type in the name of the person and the user's extension in the extension setup page, and that person is automatically added to the directory. AlexOn 6/7/06, Khaled Chehab [EMAIL PROTECTED] wrote: Please can any one help me how to make directories at [EMAIL PROTECTED] To use it from the ivr *411 Thanks * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I can hear only one way when I use nokia e-60 withX-lite
2006/6/7, Jon Schøpzinsky [EMAIL PROTECTED]: HelloBe aware that the Nokia E60, E61 and E70 does not support NAT.Just to be shure that you know that.A clever choice from Nokia, so that users has to have some local equipment from the telco.Jon What do you mean by users has to have some local equipment from the telco ?Do you think Nokia E60, E61 and E70 are appropriate for Fixed Mobile Convergence (each mobile phone being reachable at the same time from inhouse PBX and Telco's mobile network without any handover or roaming between both networks) ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] meetme public
hi all i have an asterisk working and i need to add a mettme public
service.
for example i need to download a soft (sjphone) and without any
configuration call to [EMAIL PROTECTED] (meetme) and join a conference but when
i do that i
received an error saying nomber do not exist. but if i call a extension
is work propperly.
in the extensions.conf have
exten = 411,1,Answer
exten = 411,2,Wait(1)
exten =
411,3,SetVar(CALLFILENAME=/var/spool/asterisk/monitor/${TIMESTAMP})
exten = 411,4,Monitor(wav,${TIMESTAMP},m)
exten = 411,5,Meetme(4001,qM)
exten = 411,6,Hangup
4001 is the room number
in the mmetme conf have
conf = 4001
any comments?
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SV: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite
Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work. That means, that you cant just walk down to your local airport, and use the IP part of the phone on their network. You have to have a non nat local server, to get it to run. Other than that, the phone can accept calls both from cellular network and IP network, and actuatly works quite well, both for cellular and IP traffic. But you cant do seamless handover, for example when you walk out of the building. You have two different numbers, your mobile number and your IP number And these cant automaticly be transferred. Hope this answeres your question Regards Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Olivier Krief Sendt: 7. juni 2006 16:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite 2006/6/7, Jon Schøpzinsky [EMAIL PROTECTED]: Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon What do you mean by users has to have some local equipment from the telco ? Do you think Nokia E60, E61 and E70 are appropriate for Fixed Mobile Convergence (each mobile phone being reachable at the same time from inhouse PBX and Telco's mobile network without any handover or roaming between both networks) ? Regards -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install
Did you check your mpg123 version ?, asterisk needs a specific version in order to work... - Original Message - From: Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02 AM Subject: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install I have followed the instructions provided at:http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.confincluding installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now.However, no sound is heard and I get this message from the CLI when accessing MOH:-- Started music on hold, class 'default', on channel 'Zap/19-1'-- Stoped music on hold on Zap/19-1This happens whether it's a parked call or whether I access MOH directly via:exten = 800,1,Answerexten = 800,2,MusicOnHold()Any help would be greatly appreciated.Thank you very much.Richard __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: GXP-2000 (steer clear)
Flames about GXP-2000 poor quality are frequent on this mailing list. I recently setup 80 of these (and I'm waiting for other 30...) to move whole company from a legacy Alcatel PBX to an Asterisk-only solution. At first, I tried some chinese phones (AtCom) and they were a disaster. Then I tried Grandstream phones and it was a real jump: users now are happy, I had only one RMA, quality and stability are good and I'm able to focus myself on improving Asterisk features. IMHO: +100 Euros for a phone are a theft! -- Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Prices of g729 codec
For all the noise about this noone has mentioned one important thing. We should be gratefull that we have access to G.729a in Asterisk, whatever the mechanics of the licensing. It's obvious that its a pain in the [EMAIL PROTECTED] for Digium who absolutely not making ANY on it money for their efforts. It would be really easy for them to say "no more" and it wouldn't really impact their business at all, except to reduce their headaches. This will be especially true when they introduce their new hardware based transcoding engine. Why then should they continue to deal with the per stream softwaer codec licensing? If you want access to G.729a just buy the board...the license cost withn be buried in the price and they can afford to provide support to paying customers. Again, we should be gratefull! It could very easily go away altogether. Those of you constantly complaining...this is supposed to be a open source community...don't just demand a better licensing scheme...design and implement one. That can be your contibution to the project. I'm not a code jockey or I'd have a go myself. In the interest of full disclosure, I have a small systems based upon Astlinux and a Soekris Net4801. I have 2 G.729a licenses on that box and I'd like to see Digium make the codec possibleusing the alternative C libraries that Kristian has used in Astlinux 0.4. I probably can't justify buying the hardware transcoder. And I definitely don't want them to withdraw the current codec offering.Michael GravesSr Product SpecialistPixel Power Inc[EMAIL PROTECTED]o(713) 861-4005o(800) 905-6412f(713) 864-8668c(713) 201-1262 Original Message Subject: Re: [Asterisk-Users] Prices of g729 codecFrom: "Woodoo People .pGa!" [EMAIL PROTECTED]Date: Mon, June 05, 2006 10:15 amTo: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Talk to digium about this on [EMAIL PROTECTED], they might be able to help you out there. Zoa Chris Mason (Lists) wrote: I have no problem with paying Digium the $10 for G729 licenses, everyone has to make money. It's the administration of the licenses that sucks. I experiment with different hardware a lot, and make up demo machines to install for customers with available hardware. I have to put G729 licenses on them, usually $100 each time, and when I insta ll the real hardware for the client, I can't transfer the licenses. If I scrap that machine or change the interfaces, that's a $100 loss. I believe when you buy a number of licenses, that should determine how many instances you can use, regardless of how you want to deploy them. In short, the method of enforcement is poor and leads to resentment from customers. Surely Digium can construct a better system?i think, for those of us, who would like to transfer licences from one boxto other (i mean more than 1-2 or 10), we would have to buy a hardwarebase lock (of course, i don't care about, if the lock would contactdigium once a day or so) like usb, or a dumb pci ethernet card, soif we need we can move it to other. what do you think?(sadly there is no a 7day demo licence or anything to test) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Quad T1 Card
If you can't afford to purchase both cards, then a safe bet is to simply purchase the Sangoma card since it can address more echo issues then the Digium card. Also, don't forget that the high-end A104d has more than on-board EC. It has on-board DSP handling and a 5 year warranty. Check it out: http://www.sangoma.com/datasheets/p_aft-104d-specs Having your T1 card use its muscle to process digital signals can be a luxury or it can be a necessity. I say luxury because most T1 cards that work with * simply let the server's CPU do all of the DSP work. However, in a demanding environment it might better to let the T1 card share some of the workload, allowing your CPU to handle all of the other things that CPU's are supposed to be doing. Still your call, but if this is a professional install in a mission-critical environment with significant traffic then the choice probably has been made for you already... -MC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GXP-2000 (steer clear)
Hello Mimmus, * Mimmus [EMAIL PROTECTED] [07-06-06 16:52]: At first, I tried some chinese phones (AtCom) and they were a disaster. you talking ybout this phone? http://iaxtalk.com/index.php?main_page=product_infoproducts_id=2 Has anyone some experience with this phone? Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
On Monday 05 June 2006 15:41, Andrew Kohlsmith wrote: The (current) problem is that the registration program does not ask which ethernet card you wish to bind to, nor does it look at the Asterisk config and use the MAC address of the ethernet card whose IP address is referenced in bindaddr (as an example). It grabs eth0 and runs. Has anyone tried renaming the interfaces on the box? On all my systems I rename the ethernet interfaces to more friendly names (dmz, lan, ext) so there is no ambiguity. If the license verification code is really looking for eth0 it might be possible to juggle some interface names until the USB ethernet interface shows up as eth0. On Linux, consult ifrename(8) I haven't tried it but it might work. /BAK/ -- Ben Klang Alkaloid Networks [EMAIL PROTECTED] 404.475.4850 http://projects.alkaloid.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
Hello, I have done a lot of testing on both the Digium TE406P and the Sangoma a104d and was involved in debugging both of them with Digium and Sangoma in their early releases. Since we are on a Digium-owned list right now and I don't want to be branded an enemy of Asterisk again for suggesting that you might consider buying a non-Digium product, I will mention right up front that a large portion of your purchase price from buying a Digium card will go toward keeping Asterisk development going, in fact it is how Digium makes most of their money and allows them to have dozens of programmers working full time on Asterisk. Sangoma does contribute to the Asterisk codebase, but buying a Sangoma card will not help the owner of Asterisk further improve their product at all. Now on to my recommendation. As I mentioned we have had both the Digium and Sangoma echo-cancellation cards in production for over 6 months on heavy load Asterisk servers running both 1.2.X Asterisk. Both had initial problems with drivers with the Sangoma side being fixed within a couple weeks and the Digium side being fixed by having to manually disable the hardware DTMF detection in the wct4xxp.c driver code every time I upgrade zaptel. Both of the cards do a good job at removing echo from our calls, and they both have a fairly equal effect of reducing the overall load on your system(10-20%). So performance-wise in our tests in our environment they are pretty much the same. As for the technical specs on the echo-cancellation modules used, the Sangoma card uses an Octastic chipset that is highly configurable and is one of the best telecom echo-cancellation chipsets in the industry. Is has a configurable tail length and is capable of dynamically being turned on and off as needed by it's firmware. The Digium card uses an Oki chipset that has a smaller echo tail length and is hard-coded into the firmware so you cannot change it. The other differences are just the usual differences between Digium and Sangoma cards: Digium - ready to go just loading zaptel and Asteirsk, Sangoma - must load wanpipe drivers and configure each span before using, also must recompile zaptel after installing/upgrading wanpipe driver Digium - 2 year warranty, Sangoma 5 year warranty Digium - has motherboard incompatibility list, Sangoma - guarantees functionality with all modern PCI-compliant motherboards Hope that helps, MATT--- On 6/7/06, Rich Adamson [EMAIL PROTECTED] wrote: Sean Cook wrote: One of the primary differences between the two cards is the Sangoma h/w echo canceler handles more cases of echo then do the Digium cards. Whether you need that additional coverage is 100% dependent on your specific implementation (eg, your T1/PRI provider), and not on what the list thinks about the two products. Since there are no affordable tools to truly quantify echo for each specific implementation, as a pbx engineer your toolkit should probably include both cards. Sort of like try the less expensive card and if it doesn't address your echo issues, then try the more expensive one. No offense but isn't that like saying Don't take what the list has to say about your purchase... instead you should guess and hope you get the right answer... but if you don't, gamble again and buy two cards? The list cannot guess at what level of echo you are going to incur, therefore there is no way for anyone to accurately tell you how to address issues. Both cards are quality products, but with slightly different operational characteristics. If you can't afford to purchase both cards, then a safe bet is to simply purchase the Sangoma card since it can address more echo issues then the Digium card. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: GXP-2000 (steer clear)
* Mimmus [EMAIL PROTECTED] [07-06-06 16:52]: At first, I tried some chinese phones (AtCom) and they were a disaster. you talking ybout this phone? http://iaxtalk.com/index.php?main_page=product_infoproducts_id=2 Yes DV ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk not waiting for EM Wink (I think)
Hi All,
I have a rather peculiar problem. Whenever I dial out over ZAP/g0 the
phone will just ring and ring, even if I answer the phone on the other
end. Whats strange is that the * phone will continue to ring even after
I've answered and (sometimes) hung up the dialed phone. If I make an
extension to just directly dial out on ZAP/1, its almost the same
behavior, it will continue to ring, but it will connect the call and
continue to ring. Its strange. I saw this over at digiums bug tracking
database
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=3772history=1
and I think thats my issue, but the ticket is quite old and I would've
thought they would've fixed whatever was causing it by now. I'll attach
configs and a snip of the logs. Anyone know how to fix this? (BTW, this
is not a PRI or BRI I am connecting to). Also, I've tested this using
Xten Lite and a Linksys PAP2 device, with the same results. (Both SIP)
Any help would be appreciated,
Derek
zaptel.conf:
span=1,1,0,esf,b8zs
em=1-24
loadzone = us
defaultzone=us
zapata.conf:
[channels]
language=en
context=from-pstn
signalling = em_w
rxgain=2
group = 0
channel = 1-24
/var/log/asterisk/full (snip):
Jun 7 08:15:09 DEBUG[14754] channel.c: Not copying variable SIPURI.
Jun 7 08:15:09 DEBUG[14758] app_queue.c: Device 'Zap/1' changed to
state '2' (In use) but we don't care because they're not a member of any
queue.
Jun 7 08:15:09 DEBUG[14754] chan_zap.c: Dialing '(snipped)'
Jun 7 08:15:09 DEBUG[14754] chan_zap.c: Deferring dialing...
Jun 7 08:15:09 DEBUG[13721] channel.c: Avoiding initial deadlock for
'Zap/1-1'
Jun 7 08:15:09 VERBOSE[14754] logger.c: -- Called g0/(snipped)
Jun 7 08:15:09 DEBUG[13721] devicestate.c: Changing state for Zap/1 -
state 2 (In use)
Jun 7 08:15:09 DEBUG[14759] app_queue.c: Device 'Zap/1' changed to
state '2' (In use) but we don't care because they're not a member of any
queue.
Jun 7 08:15:10 DEBUG[14754] chan_zap.c: Exception on 18, channel 1
Jun 7 08:15:10 DEBUG[14754] chan_zap.c: Got event Wink/Flash(3) on
channel 1 (index 0)
Jun 7 08:15:10 DEBUG[14754] chan_zap.c: Ignoring wink on channel 1
Jun 7 08:15:10 DEBUG[14754] chan_zap.c: Exception on 18, channel 1
Jun 7 08:15:10 DEBUG[14754] chan_zap.c: Got event Hook Transition
Complete(12) on channel 1 (index 0)
Jun 7 08:15:12 DEBUG[14754] chan_zap.c: Exception on 18, channel 1
Jun 7 08:15:12 DEBUG[14754] chan_zap.c: Got event Dial Complete(9) on
channel 1 (index 0)
Jun 7 08:15:12 DEBUG[14754] chan_zap.c: No echo cancellation requested
Jun 7 08:15:12 DEBUG[13721] channel.c: Avoiding initial deadlock for
'Zap/1-1'
Jun 7 08:15:12 DEBUG[13721] devicestate.c: Changing state for Zap/1 -
state 6 (Ringing)
Jun 7 08:15:12 DEBUG[14760] app_queue.c: Device 'Zap/1' changed to
state '6' (Ringing) but we don't care because they're not a member of
any queue.
Jun 7 08:15:25 DEBUG[13726] chan_sip.c: = Found Their Call ID:
[EMAIL PROTECTED] Their Tag dee5f8aff7d531a4o0 Our tag:
as61f0f08a
Jun 7 08:15:25 DEBUG[13726] chan_sip.c: Received CANCEL (14) -
Command in SIP CANCEL
Jun 7 08:15:25 DEBUG[14754] channel.c: Hanging up channel 'Zap/1-1'
Jun 7 08:15:25 DEBUG[14754] chan_zap.c: zt_hangup(Zap/1-1)
Jun 7 08:15:25 DEBUG[14754] chan_zap.c: Hangup: channel: 1 index = 0,
normal = 18, callwait = -1, thirdcall = -1
Jun 7 08:15:25 DEBUG[14754] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/1-1
Jun 7 08:15:25 DEBUG[14754] chan_zap.c: Updated conferencing on 1, with
0 conference users
Jun 7 08:15:25 VERBOSE[14754] logger.c: -- Hungup 'Zap/1-1'
Jun 7 08:15:25 DEBUG[14754] app_dial.c: Exiting with DIALSTATUS=CANCEL.
Jun 7 08:15:25 DEBUG[14754] app_macro.c: Spawn extension
(macro-dialout-trunk,s,14) exited non-zero on 'SIP/1040-45d7' in macro
'dialout-trunk'
Jun 7 08:15:25 DEBUG[13721] devicestate.c: Changing state for Zap/1 -
state 0 (Unknown)
Jun 7 08:15:25 DEBUG[14754] pbx.c: Spawn extension
(macro-dialout-trunk,s,14) exited non-zero on 'SIP/1040-45d7'
Jun 7 08:15:25 DEBUG[14761] app_queue.c: Device 'Zap/1' changed to
state '0' (Unknown) but we don't care because they're not a member of
any queue.
Jun 7 08:15:25 DEBUG[14754] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Jun 7 08:15:25 DEBUG[14754] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2006-06-07 08:15:09','\Vonage\
1040','1040','9(snipped)','from-internal',
'SIP/1040-45d7','Zap/1-1','Dial','ZAP/g0/(snipped)120|r',16,0,'NO
ANSWER',3,'','1149693309.30')
Jun 7 08:15:25 DEBUG[14754] pbx.c: Function result is 'Vonage 1040'
Jun 7 08:15:25 DEBUG[14754] pbx.c: Function result is '1040'
Jun 7 08:15:25 DEBUG[14754] pbx.c: Function result is '(snipped)'
Jun 7 08:15:25 DEBUG[14754] pbx.c: Function result is 'from-internal'
Jun 7 08:15:25 DEBUG[14754] pbx.c: Function result is
RE: [Asterisk-Users] Prices of g729 codec
On Wed, 2006-06-07 at 07:55 -0700, [EMAIL PROTECTED] wrote: For all the noise about this noone has mentioned one important thing. We should be gratefull that we have access to G.729a in Asterisk, whatever the mechanics of the licensing. It's obvious that its a pain in the [EMAIL PROTECTED] for Digium who absolutely not making ANY on it money for their efforts. It would be really easy for them to say no more and it wouldn't really impact their business at all, except to reduce their headaches. but they do in 2004 mark said it was one of their biggest revenue streams. Or do you mean that they dont make any money selling asterisk under their business edition line? Or maybe they dont make any money selling the hardware to people who buy it to 'support' asterisk development. I believe that cnet said they made over $10M/year in an article about an interview with mark. If $10M/year is not ANY money I would like to not make any money too. This will be especially true when they introduce their new hardware based transcoding engine. Why then should they gee just like sangoma (only sangoma anounced it first :) wonder if it will still use the zap interface and choke the system with more interrupts than required. I also wonder when asterisk will have better sangoma support so you can cut your interrupts from say 1000/sec to 50/sec. But that probably wont happen in tree. Again, we should be gratefull! It could very easily go away altogether. the codec? there will be alternatives for a licensed g729a and B (for those that want to do VAD when that is implemented) codec for asterisk. Those of you constantly complaining...this is supposed to be a open source community...don't just demand a better licensing scheme...design and implement one. That can be your contibution to the project. I'm not a code jockey or I'd have a go myself. its being done. Infact I am on the phone with some people talking about that right now. And have been for a little while (on/off for a couple months). Now you say that you arent a code monkey so you are unable to write one, but you can suggest what others should do. Specifically code something new. Hmm sounds like you just did the very thing you are complaining about. So I am lost are you complaining about your post now or what? All I have to say is that at least you can (aparently I still havent tested it with asterisk) port your digium licenses for which you paid when digium is closed but your business isnt :) and to show that I am not just suggesting a different licensing model but actually contributing here is the link to the BSD licensed code (whee its not gpl) for a trivial program and thus is my contribution. Note I have no disclaimer on file as the gpl is against my religion and as such am barred from contributing to asterisk directly. http://www.0xdecafbad.com/Remapping-function-calls.html -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install
Thank you very much for your relply. No I did not install mpg123 as the instructions at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conf for version 1.2 say the mpg123 is no longer needed. | Rurouni Alucard | [EMAIL PROTECTED] wrote: Did you check your mpg123 version ?, asterisk needs a specific version in order to work... - Original Message -From:Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02AM Subject: [Asterisk-Users] Music On Holdnot working with new 1.2.7.1 install I have followed the instructions provided at:http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.confincludinginstalling asterisk-addons-1.2. I have left musiconhold.conf as is,calm-river et al are fine for now.However, no sound is heard and I getthis message from the CLI when accessing MOH:-- Started music on hold,class 'default', on channel 'Zap/19-1'-- Stoped music on hold onZap/19-1This happens whether it's a parked call or whether I accessMOH directly via:exten = 800,1,Answerexten =800,2,MusicOnHold()Any help would be greatly appreciated.Thankyou very much.Richard __Do YouYahoo!?Tired of spam? Yahoo! Mail has the best spam protection aroundhttp://mail.yahoo.com ___--Bandwidth andColocation provided by Easynews.com --Asterisk-Users mailinglistTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Quad T1 Card
Asterisk Hater.. :) Sorry matt couldn't resist.. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Wednesday, June 07, 2006 11:20 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Quad T1 Card Hello, I have done a lot of testing on both the Digium TE406P and the Sangoma a104d and was involved in debugging both of them with Digium and Sangoma in their early releases. Since we are on a Digium-owned list right now and I don't want to be branded an enemy of Asterisk again for suggesting that you might consider buying a non-Digium product, I will mention right up front that a large portion of your purchase price from buying a Digium card will go toward keeping Asterisk development going, in fact it is how Digium makes most of their money and allows them to have dozens of programmers working full time on Asterisk. Sangoma does contribute to the Asterisk codebase, but buying a Sangoma card will not help the owner of Asterisk further improve their product at all. Now on to my recommendation. As I mentioned we have had both the Digium and Sangoma echo-cancellation cards in production for over 6 months on heavy load Asterisk servers running both 1.2.X Asterisk. Both had initial problems with drivers with the Sangoma side being fixed within a couple weeks and the Digium side being fixed by having to manually disable the hardware DTMF detection in the wct4xxp.c driver code every time I upgrade zaptel. Both of the cards do a good job at removing echo from our calls, and they both have a fairly equal effect of reducing the overall load on your system(10-20%). So performance-wise in our tests in our environment they are pretty much the same. As for the technical specs on the echo-cancellation modules used, the Sangoma card uses an Octastic chipset that is highly configurable and is one of the best telecom echo-cancellation chipsets in the industry. Is has a configurable tail length and is capable of dynamically being turned on and off as needed by it's firmware. The Digium card uses an Oki chipset that has a smaller echo tail length and is hard-coded into the firmware so you cannot change it. The other differences are just the usual differences between Digium and Sangoma cards: Digium - ready to go just loading zaptel and Asteirsk, Sangoma - must load wanpipe drivers and configure each span before using, also must recompile zaptel after installing/upgrading wanpipe driver Digium - 2 year warranty, Sangoma 5 year warranty Digium - has motherboard incompatibility list, Sangoma - guarantees functionality with all modern PCI-compliant motherboards Hope that helps, MATT--- On 6/7/06, Rich Adamson [EMAIL PROTECTED] wrote: Sean Cook wrote: One of the primary differences between the two cards is the Sangoma h/w echo canceler handles more cases of echo then do the Digium cards. Whether you need that additional coverage is 100% dependent on your specific implementation (eg, your T1/PRI provider), and not on what the list thinks about the two products. Since there are no affordable tools to truly quantify echo for each specific implementation, as a pbx engineer your toolkit should probably include both cards. Sort of like try the less expensive card and if it doesn't address your echo issues, then try the more expensive one. No offense but isn't that like saying Don't take what the list has to say about your purchase... instead you should guess and hope you get the right answer... but if you don't, gamble again and buy two cards? The list cannot guess at what level of echo you are going to incur, therefore there is no way for anyone to accurately tell you how to address issues. Both cards are quality products, but with slightly different operational characteristics. If you can't afford to purchase both cards, then a safe bet is to simply purchase the Sangoma card since it can address more echo issues then the Digium card. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be
Re: [Asterisk-Users] Prices of g729 codec
On Wed, 2006-06-07 at 11:17 -0400, Ben Klang wrote: On Monday 05 June 2006 15:41, Andrew Kohlsmith wrote: The (current) problem is that the registration program does not ask which ethernet card you wish to bind to, nor does it look at the Asterisk config and use the MAC address of the ethernet card whose IP address is referenced in bindaddr (as an example). It grabs eth0 and runs. Has anyone tried renaming the interfaces on the box? On all my systems I rename the ethernet interfaces to more friendly names (dmz, lan, ext) so there is no ambiguity. If the license verification code is really looking for eth0 it might be possible to juggle some interface names until the USB ethernet interface shows up as eth0. I havent used ifrename so I decided to install it and see, it appears to fully masq the name so that when you get the device by name it doesnt exist as the old (meaning its not an alias but a real rename). The device cant be up when this happens though, which may pose problems for some who dont want to down the interface. I dont know about the register tool, whether or not that takes more than one ethN device, but I have heard rumors that the codec itself will try eth0, eth1, etc until it gets an error back from the ioctl() saying that the device doesnt exist. If its pulling all those devices then it stands to reason that it will use them all when comparing the licenses/* files. But again I dont use the digium codec but instead the 3rd party one that as yet isnt released to the public, so I really cant say if this is true or not. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QuadBri card
Hello, I install the latest release of Asterisk, QuadBri driver. I compile al; that, copy zaptel.conf file, modify /etc/rc.d/rc.local for launch qozap... zaptel.conf: --- # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=fr defaultzone=fr span=1,1,3,ccs,ami span=2,1,3,ccs,ami span=3,1,3,ccs,ami span=4,1,3,ccs,ami bchan=1-2 dchan=3 bchan=4-5 dchan=6 bchan=7-8 dchan=9 bchan=10-11 dchan=12 zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] switchtype=euroisdn pridialplan=local prilocaldialplan=local language=fr context=from-pstn ;signalling=fxs_ks ; OLS signalling=bri_cpe_ptmp rxwink=300; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 ;Include AMP configs #include zapata_additional.conf Asterisk is OK, but when i plug my ISDN phone lines, the leds of the QuadBri card stays red! Nothing happen when i call the phone number by external line. I always have at asterisk console the message: qozap: not re-activating layer1 span1 I see my channels with zap show status if I do: less /proc/zaptel/3 (par exemple), I have the return message for each channel: DEACTIVATING What's happen?? Best regards, -- Olivier Saulnier STEGANUX 1er étage Diamecans Bel Air 03410 St Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GXP-2000 (steer clear)
Hello Mimmus, * Mimmus [EMAIL PROTECTED] [07-06-06 17:20]: Yes good to known. I played with the idea to buy one of these. You would suggest GrandStream then? Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] polycom ftp
Anydody need some access to polycom ftp server ? Harry __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Set(CDR(userfield)) Trouble
Hi, I have troubles setting the userfield in mysql ( using asterisk 1.2.8 / addons 1.2.3 ) I use this in my dialplan: exten = s,n,SetCDRUserField(SOMEVALUE) I tried also: exten = s,n,Set(CDR(userfield)=SOMEVALUE) But everytime i look at the cdr database the userfield is still empty Does anyone has a clue on how to get things working ? Thanks in advance ! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bewan phonebox
hello How can I configure a bewan phonebox with asterisk thanks issam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install
convert the moh sounfile to pcm or sln save the file to /var/lib/asterisk/moh/default set the musiconhold.conf [default]mode=filesdirectory=/var/lib/asterisk/moh/default turby@ www.canistec.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard ReinaSent: Wednesday, June 07, 2006 5:30 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install Thank you very much for your relply. No I did not install mpg123 as the instructions at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conffor version 1.2 say the mpg123 is no longer needed.| Rurouni Alucard | [EMAIL PROTECTED] wrote: Did you check your mpg123 version ?, asterisk needs a specific version in order to work... - Original Message - From: Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02 AM Subject: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install I have followed the instructions provided at:http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.confincluding installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now.However, no sound is heard and I get this message from the CLI when accessing MOH:-- Started music on hold, class 'default', on channel 'Zap/19-1'-- Stoped music on hold on Zap/19-1This happens whether it's a parked call or whether I access MOH directly via:exten = 800,1,Answerexten = 800,2,MusicOnHold()Any help would be greatly appreciated.Thank you very much.Richard __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: fine-tuning asterisk questions
On 6/6/06, M.Hockings [EMAIL PROTECTED] wrote:
William Piper wrote: For Problem #2: I'm not sure what you are asking. Perhaps post your dialplan for this
problem we will take a look. bp On 6/4/06, *M.Hockings* [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote: Problem 2) Incoming sip calls from my voip provider get rejected unless I allow anyone to connect with sip. I have an incoming route set up with
the right DID that matches the DID that asterisk picks out but it still rejects the call.Any suggestions about how to get this to work without allowing any sip connection?
MikeHi William, at the bottom of this is my extensions.conf which seems tobe the largest part of the equation for problem #2.I have not appliedany changes to try and resolve my problem #1 yet.
I think the question here is the operation of the following statement inthe [from-sip-external] section:exten = s,1,GotoIf($[${ALLOW_SIP_ANON}=yes]?from-trunk,${DID},1)
If I interpret it correctly it should go to from-trunk,1 if the freePBXallow anonymous sip connections is true and go toincoming-sip-did-value,1 if it is false ?That is should I be lookingfor something like this in the config files to understand how this would
be handled?exten=416967,1,As an aside, is there some beginners guide to understanding dial plans?My original dial plan (based on things read on voip-info.org
) was verysimple and worked as far as it was configured.I have recently gone tofreePBX to try and make the dial plan changes easier and faster howeverit adds a lot of gorp like this that I don't understand.
Thanks for any guidance on this,Mike
I have no idea about FreePBX. I thought you were trying to create something from new. I believe thatAsterisk @ homehas a list of thier own, you may want to check there.
From my personal experience, Asteirsk @ Home is really good for the AMP, but to make it work, I deleted the extensions.confand created my own then only work directly in the extensions.conf file, not AMP. Just use AMP for reports such.
I wish I could help you but I can't spend half the day trying to figure out how FreePBX works then figure out your problem.
Regards,
bp
___
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Re: [Asterisk-Users] Directory problem
Appearntly after letting Asterisk run overnight, the problem is back, from the inconsistancy of the problem I'm going to assume it's a DTMF problem, I will try working on that and see if it helps. The calls are coming in thru a Mediatrix 1204, I guess I will have to play around with the DTMF settings on the mediatrix to get it working properly, if anybody here has a Mediatrix, can they please share their DTMF settings? Thank You On 6/6/06, C F [EMAIL PROTECTED] wrote: After upgrading the problem is now gone. Thank you Kevin. On 6/6/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: - C F [EMAIL PROTECTED] wrote: Asterisk SVN-tag-1.2.1-r7367 built by root @ pbx on a i686 running Linux on 2005-12-27 19:05:04 UTC It doesn't make any sense to report a problem against a 6 month old release of Asterisk when there are newer releases available. Please review the ChangeLog files for all the releases since 1.2.1 to see if your problem has been addressed... if so, then upgrade. If not, then upgrade anyway, reproduce the problem and then file a bug report. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CISCO POE
The POE switch needs to support always on. Most switches check the device for 802.3af support before turning on power. The phones only support the CDP power activation, not 802.3af. I've used always on POE injectors from wireless access points successfully with Cisco phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of nik600 Sent: Wednesday, June 07, 2006 7:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CISCO POE many thanks for your reply i've tried to make a cable with that configuration but it seems that it doesn't work... i'm using a 7905G Cisco ip phone and an ALL0484 Switch POE thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun 7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (66ms / 500ms)Jun 7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (68ms / 500ms)Jun 7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO LAGGED! (1114ms / 500ms)Jun 7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now REACHABLE! (90ms / 500ms)Jun 7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1077ms / 500ms)Jun 7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (72ms / 500ms)Jun 7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO LAGGED! (1077ms / 500ms)Jun 7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now REACHABLE! (73ms / 500ms)As you can see, it only happens to a couple of their phones and at random times. They're behind a DSL circuit. I don't know if it's because their DSL line is going up/down. They don't necessarily claim their Internet goes down, however, they are not constantly check it.What would you (or anyone else) suggest?Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time? If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk nagios plugin
Leonimar,search check_zaptel at http://www.nagiosexchange.org, there is a simple plugin that checks alarm status on zaptel interfaces and can be used with nrpe. Sample usage: # /usr/lib/nagios/plugins/check_zaptel -s1 -s2 -s3 ZAPTEL OK: TE4/0/1 , TE4/0/2 , TE4/0/3 # /usr/lib/nagios/plugins/check_zaptel -s1 -s2 -s3 -s4 ZAPTEL Critical: TE4/0/1 , TE4/0/2 , TE4/0/3 , TE4/0/4 RED # /usr/lib/nagios/plugins/check_zaptel -s1 -vv ZAPTEL OK: Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4 Ezio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set(CDR(userfield)) Trouble
Hello, I ran into something similar and found the following in the wiki... Note : If using cdr_mysql addon make sure to set userfield=1 to in cdr_mysql.conf. If using cdr_csv, edit cdr_csv.c and (re)compile to enable the user field. This command has no effect if the user field is not enabled. See: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetCDRUserField This actually was not my problem, but it was good information. I was actually setting it and then overwriting it later on in the dialplan. Hope the information is helpfull. On 6/7/06, Tristan [EMAIL PROTECTED] wrote: Hi,I have troubles setting the userfield in mysql ( using asterisk 1.2.8 /addons 1.2.3 )I use this in my dialplan: exten = s,n,SetCDRUserField(SOMEVALUE)I tried also:exten = s,n,Set(CDR(userfield)=SOMEVALUE)But everytime i look at the cdr database the userfield is still emptyDoes anyone has a clue on how toget things working ? Thanks in advance !___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Origination that includes real support! http://www.VoIPStreet.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GXP-2000 (steer clear)
Anyone try out the Snom 300 phone yet? Seems like a decent price. On 6/7/06, Matthias Fechner [EMAIL PROTECTED] wrote: Hello Mimmus, * Mimmus [EMAIL PROTECTED] [07-06-06 17:20]: Yes good to known. I played with the idea to buy one of these. You would suggest GrandStream then? Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How-To monitor a specific channel?
Gruys, How can I record a specific channel if Monitor doesn't receive it as a parameter? Can I do a combination with the ZapBarge app? I want to record calls in some channels. Thanks in advance. Fernando Lujan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PHP UnixODBC MS SQl 2000
Hi, I have Asterisk 12.7.1 installed through [EMAIL PROTECTED] CD. and explicitly I have installed UnixODBC and FREETDS in order to access MS SQL 2000 Database which in on Windows 2003 Server on remote location. I tested connectivity through isql and tsql, both utilities are working fine. I need to access MS SQL 2000 Database through PHP. When I tired to check the connectivity through a Test PHP file I got following results: Fatal error: Call to undefined function: odbc_connect() in /var/www/html/odbctest.php on line 3 By Default PHP was configured with following switches: './configure' '--build=i686-redhat-linux-gnu' '--host=i686-redhat-linux-gnu' '--target=i386-redhat-linux-gnu' '--program-prefix=' '--prefix=/usr' '--exec-prefix=/usr' '--bindir=/usr/bin' '--sbindir=/usr/sbin' '--sysconfdir=/etc' '--datadir=/usr/share' '--includedir=/usr/include' '--libdir=/usr/lib' '--libexecdir=/usr/libexec' '--localstatedir=/var' '--sharedstatedir=/usr/com' '--mandir=/usr/share/man' '--infodir=/usr/share/info' '--cache-file=../config.cache' '--with-config-file-path=/etc' '--with-config-file-scan-dir=/etc/php.d' '--enable-force-cgi-redirect' '--disable-debug' '--enable-pic' '--disable-rpath' '--enable-inline-optimization' '--with-bz2' '--with-db4=/usr' '--with-curl' '--with-exec-dir=/usr/bin' '--with-freetype-dir=/usr' '--with-png-dir=/usr' '--with-gd=shared' '--enable-gd-native-ttf' '--without-gdbm' '--with-gettext' '--with-ncurses=shared' '--with-gmp' '--with-iconv' '--with-jpeg-dir=/usr' '--with-openssl' '--with-png' '--with-pspell' '--with-xml' '--with-expat-dir=/usr' '--with-dom=shared,/usr' '--with-dom-xslt=/usr' '--with-dom-exslt=/usr' '--with-xmlrpc=shared' '--with-pcre-regex=/usr' '--with-zlib' '--with-layout=GNU' '--enable-bcmath' '--enable-exif' '--enable-ftp' '--enable-magic-quotes' '--enable-sockets' '--enable-sysvsem' '--enable-sysvshm' '--enable-track-vars' '--enable-trans-sid' '--enable-yp' '--enable-wddx' '--with-pear=/usr/share/pear' '--with-imap=shared' '--with-imap-ssl' '--with-kerberos' '--with-ldap=shared' '--with-mysql=shared,/usr' '--with-pgsql=shared' '--with-snmp=shared,/usr' '--with-snmp=shared' '--enable-ucd-snmp-hack' '--with-unixODBC=shared,/usr' '--enable-memory-limit' '--enable-shmop' '--enable-calendar' '--enable-dbx' '--enable-dio' '--enable-mbstring=shared' '--enable-mbstr-enc-trans' '--enable-mbregex' '--with-mime-magic=/usr/share/file/magic.mime' '--with-apxs2=/usr/sbin/apxs' Please guide me what else should I need to do. Thanks Wazb ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Notice Question
On Jun 7, 2006, at 6:42 AM, Doug Crompton wrote: I get the following * Notice ocassionally and I was curious what it means and if it can safetly be ignored or corrected. Jun 7 05:40:47 NOTICE[32153]: res_musiconhold.c:511 monmp3thread: Request to schedule in the past?!?! I see that on my box when it gets busy... No negative effects apparent. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Controlling Cisco 7960 Ringtone from Asterisk
I'm trying to change the ring tone on my 7960 from the dialplan. I've tried the example on the wiki but it doesn't seem to work. Something like: exten = 3010,1,SetVar(ALERT_INFO=Bellcore-dr1) ; selects Ringer exten = 3010,2,Dial(SIP/3010,15) I'm not sure what the Bellcore-dr1 ringer is supposed to be. I've tried replacing ALERT_INFO with another ring tone I have on my system (merlin2, merlin3, etc) but I've had no luck . I'm running Asterisk 1.2.7.1. The 7960 phones have 7.4 SIP firmware loaded on them Here is what is in my RINGLIST.DAT file: R2D2r2d2.raw Meowmeow.raw Galaga galaga.raw Ahh!ahh.pcm Doh!doh.pcm Old Style ringer1.pcm Synth Low ringer2.pcm Dungeon ringer3.pcm Lightbulb ringer4.pcm Synth High ringer6.pcm Are You There M AreYouThere.raw Are You There F AreYouThereF.raw ClockShop ClockShop.raw Curley Curley.raw Drums 1 Drums1.raw Drums 2 Drums2.raw FilmScore FilmScore.raw FlintPhone FlintPhone.raw HarpSynth HarpSynth.raw Jamaica Jamaica.raw Klaxons Klaxons.raw KotoEffect KotoEffect.raw MusicBoxMusicBox.raw Neuro Neuro.raw OhnoOhno.raw Piano 1 Piano1.raw Piano 2 Piano2.raw Pop Pop.raw Pulse Pulse1.raw Saxaphone 1 Sax1.raw Saxaphone 2 Sax2.raw Asleep asleep.raw Caramba caramba.raw MayIHelpmayihelp.raw Dilbert BossSICA-dilbert-BungeeBoss.raw Dilbert Meeting SICA-dilbert-PHB.raw NyukNyukNyukNyuk.raw Merlin2 merlin2.pcm Merlin3 merlin3.pcm Merlin4 merlin4.pcm Merlin5 merlin5.pcm Merlin6 merlin6.pcm Merlin7 merlin7.pcm If any one has this working any help would be appreciated. Thanks, Jeremiah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set(CDR(userfield)) Trouble
Shame on me, that was my trouble, seems like I didn't read enough... Thanks a lot ! Lewis Agosta a crit: Hello, I ran into something similar and found the following in the wiki... Note : If using cdr_mysql addon make sure to set userfield=1 to in cdr_mysql.conf. If using cdr_csv, edit cdr_csv.c and (re)compile to enable the user field. This command has no effect if the user field is not enabled. See: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetCDRUserField This actually was not my problem, but it was good information. I was actually setting it and then overwriting it later on in the dialplan. Hope the information is helpfull. On 6/7/06, Tristan [EMAIL PROTECTED] wrote: Hi, I have troubles setting the userfield in mysql ( using asterisk 1.2.8 / addons 1.2.3 ) I use this in my dialplan: exten = s,n,SetCDRUserField(SOMEVALUE) I tried also: exten = s,n,Set(CDR(userfield)=SOMEVALUE) But everytime i look at the cdr database the userfield is still empty Does anyone has a clue on how toget things working ? Thanks in advance ! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Origination that includes real support! http://www.VoIPStreet.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Block access to [EMAIL PROTECTED]
i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users without any registration in the asterisk. how to block this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QuadBri card
On Wed, Jun 07, 2006 at 05:44:16PM +0200, Olivier Saulnier wrote: Hello, I install the latest release of Asterisk, QuadBri driver. I compile al; that, copy zaptel.conf file, modify /etc/rc.d/rc.local for launch qozap... Bad place. rc.local is just about the last place in the init sequence to be run. After Asterisk is started. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite
On Jun 7, 2006, at 7:35 AM, Jon Schøpzinsky wrote: x-tad-smallerHello Olivier/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerIve been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work./x-tad-smallerx-tad-smallerThat means, that you cant just walk down to your local airport, and use the IP part of the phone on their network./x-tad-smallerx-tad-smallerYou have to have a non nat local server, to get it to run./x-tad-smallerx-tad-smallerOther than that, the phone can accept calls both from cellular network and IP network, and actuatly works quite well, both for cellular and IP traffic./x-tad-smallerx-tad-smallerBut you cant do seamless handover, for example when you walk out of the building. You have two different numbers, your mobile number and your IP number/x-tad-smallerx-tad-smaller And these cant automaticly be transferred./x-tad-smallerx-tad-smaller /x-tad-smaller huh, well that makes it pretty useless. I wonder if this is really so? I hope it's fixed soon... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
Michael Collins wrote: If you can't afford to purchase both cards, then a safe bet is to simply purchase the Sangoma card since it can address more echo issues then the Digium card. Also, don't forget that the high-end A104d has more than on-board EC. It has on-board DSP handling and a 5 year warranty. Check it out: What does the onboard DSP do when used with Asterisk? Did Digium or someone put code inside Asterisk to hand off the processing/transcoding to a Sangoma card? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set(CDR(userfield)) Trouble
Tristan wrote: Hi, I have troubles setting the userfield in mysql ( using asterisk 1.2.8 / addons 1.2.3 ) I use this in my dialplan: exten = s,n,SetCDRUserField(SOMEVALUE) I tried also: exten = s,n,Set(CDR(userfield)=SOMEVALUE) But everytime i look at the cdr database the userfield is still empty Does anyone has a clue on how to get things working ? Make sure you have userfield=1 in your cdr_mysql.conf -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install
if on freebsd..stop asteriskkillall mpg123cd /usr/ports/audio/madplay/make make installedit musiconhold.conf[default]mode=customdirectory=/usr/local/share/asterisk/mohmp3 application=/usr/local/bin/madplay -Q -o raw:- --mono -R 8000 -a -12then restart asterisk.. Mikehttp://www.theclubvoip.com On 6/7/06, turby [EMAIL PROTECTED] wrote: convert the moh sounfile to pcm or sln save the file to /var/lib/asterisk/moh/default set the musiconhold.conf [default]mode=filesdirectory=/var/lib/asterisk/moh/default turby@ www.canistec.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Richard ReinaSent: Wednesday, June 07, 2006 5:30 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install Thank you very much for your relply. No I did not install mpg123 as the instructions at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conffor version 1.2 say the mpg123 is no longer needed.| Rurouni Alucard | [EMAIL PROTECTED] wrote: Did you check your mpg123 version ?, asterisk needs a specific version in order to work... - Original Message - From: Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02 AM Subject: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install I have followed the instructions provided at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.confincluding installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now.However, no sound is heard and I get this message from the CLI when accessing MOH:-- Started music on hold, class 'default', on channel 'Zap/19-1'-- Stoped music on hold on Zap/19-1This happens whether it's a parked call or whether I access MOH directly via:exten = 800,1,Answerexten = 800,2,MusicOnHold()Any help would be greatly appreciated.Thank you very much.Richard __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Supporter needed
Hi, I'm looking for a great tech support person to take over the admin of our asterisk system. If you are a networking person as well, with some experience in firewalls and desktop support even better. The system is a multi-group system with IVR, Follow-me dialing, voicemail, and conferencing. Multiple SIP providers are in use. If you feel you can help us, or can recommend someone that can - please let me know. Thanks We are located in Menlo Park, CA. /S ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog Line Static and Low Volume
Have a customer running a 3rd party PBX implementation based on Asterisk, not utilizing SIP inbound and outbound calls I believe are coming through a Digium TDM2402B. They are utilizing Polycom phones. They are experiencing frequent static on the line, and overall insufficient volume on conversations. They are in a bit of a rural area, I was curious if anyone thinks it could be an issue with their POTS provider, or if it is PBX related? Would adding echo cancellation perhaps alleviate static on the line, I did not think it would, just looking for some feedback. They have tweaked the gain settings and this has not produced any meaningful improvements. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP UnixODBC MS SQl 2000
Not that this is particularly an Asterisk problem, but make sure unixodbc is listed when you do a phpinfo(); , also, you might want to make sure you have extension=unixodbc.so in your php.ini since you're compiling it as a shared module. Hope that helps a little bit. Wasif wrote: Hi, I have Asterisk 12.7.1 installed through [EMAIL PROTECTED] CD. and explicitly I have installed UnixODBC and FREETDS in order to access MS SQL 2000 Database which in on Windows 2003 Server on remote location. I tested connectivity through isql and tsql, both utilities are working fine. I need to access MS SQL 2000 Database through PHP. When I tired to check the connectivity through a Test PHP file I got following results: Fatal error: Call to undefined function: odbc_connect() in /var/www/html/odbctest.php on line 3 By Default PHP was configured with following switches: './configure' '--build=i686-redhat-linux-gnu' '--host=i686-redhat-linux-gnu' '--target=i386-redhat-linux-gnu' '--program-prefix=' '--prefix=/usr' '--exec-prefix=/usr' '--bindir=/usr/bin' '--sbindir=/usr/sbin' '--sysconfdir=/etc' '--datadir=/usr/share' '--includedir=/usr/include' '--libdir=/usr/lib' '--libexecdir=/usr/libexec' '--localstatedir=/var' '--sharedstatedir=/usr/com' '--mandir=/usr/share/man' '--infodir=/usr/share/info' '--cache-file=../config.cache' '--with-config-file-path=/etc' '--with-config-file-scan-dir=/etc/php.d' '--enable-force-cgi-redirect' '--disable-debug' '--enable-pic' '--disable-rpath' '--enable-inline-optimization' '--with-bz2' '--with-db4=/usr' '--with-curl' '--with-exec-dir=/usr/bin' '--with-freetype-dir=/usr' '--with-png-dir=/usr' '--with-gd=shared' '--enable-gd-native-ttf' '--without-gdbm' '--with-gettext' '--with-ncurses=shared' '--with-gmp' '--with-iconv' '--with-jpeg-dir=/usr' '--with-openssl' '--with-png' '--with-pspell' '--with-xml' '--with-expat-dir=/usr' '--with-dom=shared,/usr' '--with-dom-xslt=/usr' '--with-dom-exslt=/usr' '--with-xmlrpc=shared' '--with-pcre-regex=/usr' '--with-zlib' '--with-layout=GNU' '--enable-bcmath' '--enable-exif' '--enable-ftp' '--enable-magic-quotes' '--enable-sockets' '--enable-sysvsem' '--enable-sysvshm' '--enable-track-vars' '--enable-trans-sid' '--enable-yp' '--enable-wddx' '--with-pear=/usr/share/pear' '--with-imap=shared' '--with-imap-ssl' '--with-kerberos' '--with-ldap=shared' '--with-mysql=shared,/usr' '--with-pgsql=shared' '--with-snmp=shared,/usr' '--with-snmp=shared' '--enable-ucd-snmp-hack' '--with-unixODBC=shared,/usr' '--enable-memory-limit' '--enable-shmop' '--enable-calendar' '--enable-dbx' '--enable-dio' '--enable-mbstring=shared' '--enable-mbstr-enc-trans' '--enable-mbregex' '--with-mime-magic=/usr/share/file/magic.mime' '--with-apxs2=/usr/sbin/apxs' Please guide me what else should I need to do. Thanks Wazb ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:44870975154201497913098! -- Derek Fedel Director of Network Development ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
What specifically were the voice quality complaints about the spa-841 phones? The only thing I have noticed is calls can be louder than expected. What else have you seen? Daniel Salama wrote: They don't all go down at the same time, or at least, my client hasn't noticed. I just added the qualify option. Let's see how that goes. As for the SPA-841, I have a client with a few of them and he cannot stop complaining about the bad audio quality. I replace a couple with a PAP-2 and another one with the GXP-2000 and he claims the quality to be incredibly better for both the PAP2 and the GXP-2000. He hasn't complained about the problems I mentioned on the GXP-2000 - yet :) Thanks, Daniel On Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote: Do you have multiple phones going down at the same time? If so, monitor them with qualify=500 in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. Also I have heard some phones have trouble with broadcast packets (at least this has been said about the spa-841 on the wiki). You should strongly consider putting them on a separate vlan to avoid any issues like that. In the future, for phones under $100 then look at the spa-841 phones. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set(CDR(userfield)) Trouble
Glad I could help. Cheers. On 6/7/06, Tristan [EMAIL PROTECTED] wrote: Shame on me, that was my trouble, seems like I didn't read enough...Thanks a lot !Lewis Agosta a écrit: Hello, I ran into something similar and found the following in the wiki... Note : If using cdr_mysql addon make sure to set userfield=1 to in cdr_mysql.conf. If using cdr_csv, edit cdr_csv.c and (re)compile to enable the user field. This command has no effect if the user field is not enabled. See: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetCDRUserField This actually was not my problem, but it was good information. I was actually setting it and then overwriting it later on in the dialplan. Hope the information is helpfull. On 6/7/06, Tristan [EMAIL PROTECTED] wrote: Hi,I have troubles setting the userfield in mysql ( using asterisk 1.2.8 /addons 1.2.3 )I use this in my dialplan: exten = s,n,SetCDRUserField(SOMEVALUE)I tried also:exten = s,n,Set(CDR(userfield)=SOMEVALUE)But everytime i look at the cdr database the userfield is still empty Does anyone has a clue on how toget things working ? Thanks in advance !___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Origination that includes real support! http://www.VoIPStreet.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Origination that includes real support!http://www.VoIPStreet.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Block access to [EMAIL PROTECTED]
Pietro U wrote: i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users without any registration in the asterisk. how to block this? Point your default value in sip.conf to an empty context. Florian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Block access to [EMAIL PROTECTED]
In your sip.conf or iax.conf you need to change the default context to something that will not interact with your main dialplan. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d:+1.813.864.3164 SIP URI: [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pietro U Sent: Wednesday, June 07, 2006 1:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Block access to [EMAIL PROTECTED] i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users without any registration in the asterisk. how to block this? This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Controlling Cisco 7960 Ringtone from Asterisk
Jeremiah Millay wrote: I'm trying to change the ring tone on my 7960 from the dialplan. I've tried the example on the wiki but it doesn't seem to work. Something like: exten = 3010,1,SetVar(ALERT_INFO=Bellcore-dr1) ; selects Ringer exten = 3010,2,Dial(SIP/3010,15) Try something like this: exten = 3020,1,Set(_ALERT_INFO=bellcore-r3) ; selects Ringer without the and . The above works with svn trunk, not sure about the format for stable releases. The default 7960 doesn't really support much in terms of different ringers although you supposedly can load different ring sounds via your tftp server files (which appears to be some of them below). I'm not sure what the Bellcore-dr1 ringer is supposed to be. I've tried replacing ALERT_INFO with another ring tone I have on my system (merlin2, merlin3, etc) but I've had no luck . I'm running Asterisk 1.2.7.1. The 7960 phones have 7.4 SIP firmware loaded on them Here is what is in my RINGLIST.DAT file: R2D2r2d2.raw Meowmeow.raw Galaga galaga.raw Ahh!ahh.pcm Doh!doh.pcm Old Style ringer1.pcm Synth Low ringer2.pcm Dungeon ringer3.pcm Lightbulb ringer4.pcm Synth High ringer6.pcm Are You There M AreYouThere.raw Are You There F AreYouThereF.raw ClockShop ClockShop.raw Curley Curley.raw Drums 1 Drums1.raw Drums 2 Drums2.raw FilmScore FilmScore.raw FlintPhone FlintPhone.raw HarpSynth HarpSynth.raw Jamaica Jamaica.raw Klaxons Klaxons.raw KotoEffect KotoEffect.raw MusicBoxMusicBox.raw Neuro Neuro.raw OhnoOhno.raw Piano 1 Piano1.raw Piano 2 Piano2.raw Pop Pop.raw Pulse Pulse1.raw Saxaphone 1 Sax1.raw Saxaphone 2 Sax2.raw Asleep asleep.raw Caramba caramba.raw MayIHelpmayihelp.raw Dilbert BossSICA-dilbert-BungeeBoss.raw Dilbert Meeting SICA-dilbert-PHB.raw NyukNyukNyukNyuk.raw Merlin2 merlin2.pcm Merlin3 merlin3.pcm Merlin4 merlin4.pcm Merlin5 merlin5.pcm Merlin6 merlin6.pcm Merlin7 merlin7.pcm If any one has this working any help would be appreciated. Thanks, Jeremiah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New York Times article on VoIP Hacker
http://www.nytimes.com/2006/06/07/technology/07cnd-voice.html?hpex=1149739200en=0f01d0becf766f0bei=5094partner=homepage Free to read, but you have to sign up. Anyone know the details of this caper? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Block access to [EMAIL PROTECTED]
On Wed, 2006-06-07 at 14:06 -0300, Pietro U wrote: i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users without any registration in the asterisk. how to block this? Remove the guest user from sip.conf and iax.conf -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog Line Static and Low Volume
On Jun 7, 2006, at 10:22 AM, Cory Andrews wrote: x-tad-smallerHave a customer running a 3/x-tad-smallerx-tad-smallerrd/x-tad-smallerx-tad-smaller party PBX implementation based on Asterisk, not utilizing SIP inbound and outbound calls I believe are coming through a Digium TDM2402B. They are utilizing Polycom phones. They are experiencing frequent static on the line, and overall insufficient volume on conversations./x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerThey are in a bit of a rural area, I was curious if anyone thinks it could be an issue with their POTS provider, or if it is PBX related? Would adding echo cancellation perhaps alleviate static on the line, I did not think it would, just looking for some feedback. They have tweaked the gain settings and this has not produced any meaningful improvements./x-tad-smallerx-tad-smaller /x-tad-smallerDefinitely sounds like it could be a PSTN issue. I also have seen both those circumstances (static and low volume) with one of my ata's (AG168V) which was rectified by a firmware update... Marty PS I don't think an Echo can. will help those issues at all. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
The complete opposite. The user complaints that either they cannot hear the remote party well or the remote party cannot hear them well. Sometimes it works and sometimes the volume is very low and that's why they cannot hear.- DanielOn Jun 7, 2006, at 1:35 PM, Mike Fedyk wrote:What specifically were the voice quality complaints about the spa-841 phones? The only thing I have noticed is calls can be louder than expected. What else have you seen? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Block access to [EMAIL PROTECTED]
thanks! it [EMAIL PROTECTED]On 6/7/06, Florian Overkamp [EMAIL PROTECTED] wrote: Pietro U wrote: i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users without any registration in the asterisk. how to block this?Point your default value in sip.conf to an empty context.Florian___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
